The present invention relates to a mobile terminal, radio communication apparatus and radio communication method in a radio communication system that performs communication from multiple antennas, wherein according to the present invention a real-time communication (RT communication) is performed using a main antenna and other antennas that have no correlation with the main antenna to realize diversity transmission, thereby RT communication gain is obtained and the error rate in RT communication is lowered.
In an Internet service that uses a mobile communication system, a packet transmission that efficiently transmits signals having various qualities and transmission rates is very suitable. A characteristic of packet transmission is that signals are transmitted only when there is data from each user, and channels can be shared by a plurality of users, so radio resources can be used efficiently. Also, adaptive modulation, scheduling and retransmission are used as methods for performing transmission even more efficiently.
(a) Adaptive Modulation Method
The characteristic of radio propagation path is constantly changing, so it becomes necessary to transmit signals according to the state of the propagation path. As an example, is a method of controlling the transmission power. When the state of the propagation path is poor, the reception quality at a receiving station is maintained at a certain level by increasing the transmission power. However, in this method, the transmission power changes, so it is feasible that the interference characteristic will change with respect to another receiving station or adjacent cell.
Therefore, as another way of thinking is a method of keeping the transmission power constant and changing the modulation parameters (data modulation method, encoding rate, etc.) in accordance to the state of the propagation path. This method is called adaptive modulation (AMC: Adaptive Modulation Control). The data is generally modulated by various multi-value modulation methods (ex. binary-, QPSK-, 16QAM-, 64QAM-modulation method) and error correction is performed. The larger the value in the multi-value modulation method, and the closer the encoding rate R in error correction is to 1, the amount of data that is sent at one time becomes larger, and therefore the tolerance to transmission error becomes weak. When the state of the propagation path is good, by increasing the value in the multi-value modulation method and bringing the encoding rate nearer to 1, the amount of transmission data and the transmission throughput are increased. On the other hand, when the state of the propagation path is poor, by decreasing the value in the multi-value modulation method and making the encoding rate small, it is possible to decrease the amount of transmission data and to prevent an increase in transmission error rate. In a system that performs spreading frequency of data, such as in the code division multiple access method (CDMA), the signal spreading rate (also called process gain) can be used as the modulation parameter. By changing the modulation parameter according to the state of the propagation path, signal transmission that matches the transmission propagation state is possible, and as a result a rise in transmission error rate is suppressed and efficient transmission becomes possible.
(b) Scheduler
In a mobile communication system, when performing signal transmission to a plurality of users in a cell, it becomes important that radio resources are efficiently assigned to each user. In other words, at what time, by which channel, by what power and how long of a packet length the packet of a user is transmitted must be determined. The unit that performs this work is called a scheduler. In a scheduler, radio resources are assigned to each user based on various information. This information may include the propagation path state for each user, the priority of the users, frequency of the occurrence of data, the amount of data, and the like, with a different type of information being used according to the system. Also, which part of the radio resources and by which standard the resources are to be assigned differs according to the system. As a prior standard for assigning resources is PF (Proportional Fairness) of HSDPA (High Speed Downlink Packet Access) that is used in W-CDMA or the like. PF is a technique of noticing fluctuations in the propagation path due to fading and selecting a moment with little propagation path loss for transmission, while at the same time maintaining equivalent transmission opportunity. As the number of connected users increases, the transmission time that is assigned to a user decreases, however, it becomes more possible to select only moments of little propagation path loss for transmission, and thus it is possible to improve the throughput of the transmitting station. The gain that is selected only when there is a good state of this fading is called MUD (Multi-User Diversity) of the scheduler. In comparison with a RR (Round Robin) that simply assigns the transmission opportunity, PF is able to obtain MUD gain and greatly improve throughput.
FIG. 29 is a drawing for explaining assignment from the scheduler, where (A) shows assignment when there are two users, and (B) shows assignment when there are three users, with SINR at UE on the vertical axis being the reception SINR (Signal to Interference Noise Ratio) that is measured by the terminal, and Tx User along the horizontal axis being the destination user that is selected by the transmitting station based on that measured value. In comparing (A) and (B), when the number of users becomes 3, it can be seen that the period that transmission is assigned decreases, however, the transmission is executed using a better propagation state.
(c) Retransmission Method
There is a retransmission method ARQ (Automatic Repeat request) for retransmitting a packet for which transmission failed. The receiving station determines whether information of a received packet is accurately decoded, and notifies the transmitting side whether transmission is succeeded or failed (ACK/NACK). The transmitting station stores the data information of transmitted packets in a buffer, and when there is a notification that transmission failed, retransmits that packet. When there is notification that a packet was received successfully, the transmitting station deletes that packet data from the buffer. When performing communication such as Internet or data communication, a certain amount of delay is allowed, however accuracy is desired. In the case of this kind of traffic, by increasing the maximum number of retransmissions, it is possible to perform communication having little packet deletion. On the other hand, in the case of traffic during real-time communication (RT communication) such as a phone call, a certain amount of packet deletion is allowed, however, the maximum number of retransmissions is set to ‘0’. In order to further improve the reception quality of a retransmitted signal, there is a retransmission combination method HARQ (Hybrid Automatic Repeat reQuest) that combines data on the receiving side. On the receiving side, when reception fails, data of the packet that contains error is stored in a buffer. When the retransmitted packet is received, it is combined with the data stored in the buffer. By combining data, reception quality is improved, and as the number of retransmissions increases, the degree of improvement becomes high and the success rate of packet reception increases.
In a packet transmission system in mobile communication, by using the techniques described above, it is possible to use radio resources efficiently, to perform packet transmission that is suitable for each user or traffic characteristics, and to perform packet transmission that is suitable for the objective of the system provider.
(d) Conventional Packet Transmission System
Transmission Side Signal Modulation Unit and Receiving Side Signal Demodulation Unit
FIG. 30 is a drawing showing an example of the construction of a transmission side signal modulation unit in a conventional packet transmission system, and FIG. 31 is a drawing showing an example of the construction of a receiving side signal demodulation unit.
In the transmission side signal modulation unit shown in FIG. 30, the modulation method, encoding rate and spreading rate are designated by the modulation parameters. Error correction and encoding is performed for the transmission data by a turbo encoding unit 3a using turbo code, for example. In the turbo encoding unit, the encoding rate is fixed (for example, R=⅓). In a punctured encoding unit 3b, several punctured code patterns are used to achieve a requested encoding rate (for example, R=¾). In a data modulation unit 3c, data modulation is performed according to a modulation method (multi-value modulation method). Typically, the modulation method is QPSK, 16QAM, 64QAM, etc. In a spreading unit 3d, the input signal is spread according to the spreading rate. In spreading, there is a method of spreading in the time direction and a method of spreading in the frequency direction.
As shown in FIG. 31, the receiving side signal demodulation unit has construction that performs retransmission combination before punctured decoding ((A) of FIG. 31), or has construction that performs retransmission combination after punctured decoding ((B) of FIG. 31). In FIG. 31, an inverse spreading unit 4a performs inverse spreading according to the spreading rate. Next, a data demodulation unit 4b performs data demodulation according to a modulation method. In the case of a retransmitted packet, a retransmission combination unit 4c performs combination processing with the same packet data received before. By doing this, it is possible to obtain higher reception quality. As described above, for retransmission combination there are two constructions, in the first construction the combination is performed before punctured decoding, and in the second construction the combination is performed after punctured decoding. A punctured decoding unit 4d performs punctured decoding according to the encoding rate, and a turbo decoding unit 4e performs turbo decoding.
Punctured Decoding
FIG. 32 is a drawing showing the construction of the punctured decoding unit 4d, and corresponds to the punctured decoding unit shown in (B) of FIG. 31. The punctured decoding unit 4d comprises a punctured code pattern generation unit 5a that generates a punctured code pattern PCP that corresponds to the encoding rate, and a data buffer 5b. The signal RD after data demodulation is stored in the data buffer 5b for the number of codes ‘1’ in the punctured code pattern PCP. The punctured code pattern PCP differs according to encoding rate, and so the buffer length also differs. FIG. 33 shows an example of a buffer length of 4 (the number of ‘1’s in the punctured code pattern PCP is 4). The stored signal RD is written to the positions of code ‘1’ of the punctured code pattern PCP, to output the signal RD′ after punctured decoding.
Construction Of the Retransmission Combination Unit on the Receiving Side
FIG. 34 is a drawing showing the construction of the retransmission combination unit on the receiving side. A buffer unit 6a saves the packet for which transmission failed together with the packet number. A buffer data extraction unit 6b makes reference to the packet number and obtains the packet data that is to be combined with the retransmitted packet from the buffer 6a. In the case where a received packet is not a retransmitted packet, or in other words, when the packet is a new packet, a retransmission combination unit 4c lets the packet pass as is and inputs it to a turbo decoding unit 4e, and in the case where the packet is a retransmitted packet, the retransmission combination unit 4c combines that packet with packet data read from the buffer 6a and inputs the result to the turbo decoding unit 4e. The turbo decoding unit 4e performs turbo decoding on the input data, and a CRC check unit 4f uses the decoded data to execute a CRC check operation and checks whether there is any error in the decoded data, and when there is no error in the data, outputs that data as received data and generates an ACK signal, and when there is error, generates a NACK signal. When a retransmitted packet is received and an ACK signal is generated, a data/information storage unit 6c deletes that packet from the buffer 6a, and when a NACK signal is received, stores the retransmitted combined data in the buffer 6a together with the packet number.
Signal Modulation/Signal Demodulation
FIG. 35 shows an example of signal modulation on the transmitting side. Here, the modulation method is 16QAM, and the encoding rate R is ¾. Taking the transmission data to be A, 6 bit data A1 to A6 is considered. When the encoding rate of turbo encoding is taken to be ⅓, the encoded data becomes B1 to B18. In a punctured code pattern PCP for an encoding rate of ¾, of the 18 bits, the 8 bits becomes ‘1’. The data B1 to B7 and B16 corresponding to the bit position of the pattern PCP where code is ‘1’ becomes the data after punctured encoding, and is output as C1 to C8 (rate matching). The original 6 bit data becomes 8 bit data, so an encoding rate of ¾ is accomplished. In data modulation, 16QAM modulation is performed, and the data C1 to C8 becomes the data E1, E2. The data E1, E2 is then spread according to the spreading rate.
FIG. 36 shows an example of signal demodulation on the receiving side. The flow is the opposite of that shown in FIG. 35. In punctured decoding, data C1 to C8 is written at bit positions of the punctured code pattern where code is ‘1’, and as a result turbo code with an encoding rate of ⅓ is obtained (de-rate matching). By performing turbo decoding on the data after punctured decoding, the original 6-bit data A1 to A6 is decoded.
Retransmission Combination
Chase combination and IR combination are typically used as retransmission combination. Here, these two methods will be explained. FIG. 37 is a drawing that explains Chase combination, where (a) of FIG. 37 is a drawing that explains Chase combination before punctured decoding, and (b) of FIG. 37 is a drawing that explains Chase combination after punctured decoding. The reference numbers will be used according to the example described above.
In combination before punctured decoding, the decoded data C1 to C8 is combined with the data C1(b) to C8(b) in the buffer 6a as shown in (a) of FIG. 37, and for example is combined at a maximum ratio. The retransmitted combined 8 bit data is substituted into the positions (B1 to B7, B16) where the code of punctured code packet PCP (see FIG. 36) is ‘1’, after which punctured decoding is performed and input to the turbo decoding unit 4e. 
In combination after punctured decoding, after the decoded data C1 to C8 is substituted into the positions (B1 to B7, B16) where the code of punctured code packet PCP (see FIG. 36) is ‘1’ as shown in (b) of FIG. 37, the data is combined with the data B1(b) to B18(b) inside the buffer 6a. In (a) and (b) of FIG. 37, the construction of the buffer is different, however the effect is the same.
FIG. 38 is a drawing that explains IR combination, where (a) of FIG. 38 is a drawing that explains IR combination before punctured decoding, and (b) of FIG. 38 is a drawing that explains IR combination after punctured decoding.
In IR combination, encoding is performed by a different punctured code pattern for each retransmission. Here, the number of patterns is taken to be two. When two patterns are used, the pattern used in the first transmission is different from that used in the first retransmission. In the second retransmission, the same pattern as the first transmission is used. Data combination is performed only when using the same pattern.
In combination before punctured decoding as shown in (a) of FIG. 38, punctured encoding is performed on the data of the first transmission and data of the second, fourth and sixth retransmissions using the same pattern PCP (see (a) of FIG. 39), so data C1 to C8 is combined with the data stored in a first buffer 6a-1, and then stored again in the buffer 6a-1. Punctured encoding is performed for the data of the first, third and fifth retransmission using a punctured code pattern PCP′ that is different from the pattern PCP (see (b) of FIG. 39), so that data D1 to D8 is combined with data stored in a different buffer 6a-2, and then stored again in the buffer 6a-2. Through punctured decoding, each of the combined data C1 to C8 and the combined data D1 to D8 is respectively written at bit positions of each punctured code pattern PCP, PCP′ where code is ‘1’ and puncture decoded. Here, Ci indicates the data of the first pattern, and Di indicates the data of the second pattern. The puncture decoded data is then input to the turbo decoding unit and turbo decoded.
In combination after punctured decoding shown in (b) of FIG. 38, after each retransmission, data is puncture decoded using each pattern PCP, PCP′ alternatively and then combined with the data in the buffer 6a. Here, only data having the same pattern are combined. Data C1 to C8 that are decoded using the first pattern PCP in the first transmission and the second, fourth, sixth . . . retransmissions, are combined with data (B1(b), B2(b) to B7(b), B16(b)) of corresponding positions in the buffer 6a. Data D1 to D8 that are decoded using the second pattern in the first, third, fifth . . . retransmissions are combined with data (B8(b) to B11(b), B13(b) to B14(b), B17(b) to B18(b)) of corresponding positions in the buffer 6a. The puncture decoded data is then input to the following turbo decoding unit and turbo decoded.
(e) MIMO Multiplexed Transmission
As a method for greatly improving the throughput of one on one communication there is MIMO (Multi Input Multi Output) multiplexed transmission.
FIG. 40 is a drawing showing the construction of a MIMO multiplexed transmission system, where TRX is a transmitting station, and REC is a receiving station. The same number of data streams D0 to DM−1 as the number of transmission antennas M go through processing such as data modulation, D/A conversion, orthogonal modulation, frequency UP conversion, and the like by respective transmission apparatuses TRX0, to TRXM−1, and are transmitted from respective transmission antennas ATT0 to ATTM−1. Signals that are transmitted from antennas ATT0 to ATTM−1, that are arranged so that they have no correlation one another pass through independent fading channels hnm (m=0 to M−1, n=0 to N−1), and after being spatially multiplexed, are received by N number of reception antennas ATR0 to ATRN−1. Signals that are received by the reception antennas undergo processing such as frequency DOWN conversion, orthogonal detection, A/D conversion and the like by receiving apparatuses REC0 to RECN−1, to generate received data streams y0 to yN−1. Each of the received data streams are in a multiplexed form of M number of transmission data streams, so in a data processing unit DPU, by performing signal processing on all of the received data streams, the transmission data streams D0 to DM−1 can be separated out and reproduced.
As algorithms for processing signals to separate out transmission data streams D0 to DM−1 from the received signals there are linear algorithms such as ZF (Zero-Forcing) or MMSE that use an inverse matrix of a channel correlation matrix (See A. van Zelst, “Space Division Multiplexing Algorithms”, 10th Mediterranean Electrotechnical Conference 2000, MELECON 2000, Cyprus, May 2000, Vol. 3, pp. 1218-1221) and there are non-linear algorithms such as BLAST (Bell Laboratories Layered Space-Time) (See P. W. Wolniansky, G. J. Foschini, G. D. Golden, R. A. Valenzuela “V-BLAST: An Architecture for Realizing Very High Data Rates Over the Rich-Scattering Wireless Channel”, Proc. 1998 Int. Symp. On Advanced Radio Technologies, Boulder, Colo., 9-11 Sep. 1998). Methods such as MLD (Maximum Likelihood Decoding) are also known that do not use inverse matrix calculation of a correlation matrix (See Geert Awater, Allert. van Zelst and Richard. van Nee, “Reduced Complexity Space Division Multiplexing Receivers,” in proceedings IEEE VTC 2000, Tokyo, Japan, May 15-18, 2000, vol. 2, pp. 1070-1074). The MLD algorithm will be explained below. By expressing transmission data streams by an M-dimensional complex matrix and received data streams by an N-dimensional complex matrix, the following relationship is formed.
      Y    =          H      ·      D            H    =          [                                                                                                        h                    00                                    ·                                      h                                          01                      ⁢                                                                                                                                          ⁢                …                ⁢                                                                  ⁢                                  h                                                            0                      ⁢                      M                                        -                    1                                                              ⁢                                                                                                                                      h                                  10                  ⁢                                                                                                    ⁢              …              ⁢                                                          ⁢                              h                                                      1                    ⁢                    M                                    -                  1                                                                                          ⋯                                                                              h                                                      N                    -                    10                                    ⁢                                                                                                    ⁢              …              ⁢                                                          ⁢                              h                                  N                  -                                      1                    ⁢                    M                                    -                  1                                                                        ]            D    =                  [                                            D              0                        ·                          D                              1                ⁢                                                                                                ⁢          …          ⁢                                          ⁢                      D                          M              -              1                                      ]            T            Y    =                  [                                            y              0                        ·                          y                              1                ⁢                                                                                                ⁢          …          ⁢                                          ⁢                      y                          N              -              1                                      ]            T      
The MLD algorithm is a method that does not use inverse matrix calculation of a correlation matrix, and a transmission data stream (transmission vector) D is estimated by the following equation.{circumflex over (D)}=arg min∥Y−H·D∥2 
Here, by taking the number of signal point arrangements of modulated data that are input to each of M number of antennas as Q, there exits QM number of combinations of transmission vectors D. In QPSK, Q=4, in 16QAM, Q=16, and in 64QAM, Q=64. The MLD algorithm is a method in which QM number of transmission vector candidates (replicas) are generated, and the calculation of the equation above is performed, and as a result, the smallest replica is estimated to be input data.
Even in MIMO multiplexed transmission, a scheduler adopting a PF is being investigated as a method for improving throughput while maintaining equivalent transmission opportunity among a plurality of users. These investigated techniques focus on improvement of throughput, and symbols for which error occurs are compensated for by retransmission techniques such as retransmission symbol combination type HARQ.
(f) RT (Real-Time) Communication
As future technology, VoIP (Voice over IP) is currently being considered. The VoIP is a kind of voice telephone real-time communication in IP network and does not use a line switching apparatus. That is, VoIP is a method that divides up compressed encoded audio data into packets and transmits the packets, and performs communication via an Internet network router. In addition, VoIP is not a connection type but connectionless type, as well as it provides flexibility of avoiding trouble and maintenance is relatively easy, so it is a very powerful technique. Another example of RT communication is online games and the like.
(g) Problems
In RT communication, it is necessary to perform communication within a fixed period of time, and retransmission delay due to HARQ that is performed when an error in a packet occurs is not allowed. The PF technique in one-on-one communication that includes the MIMO multiplexed transmission premises retransmission technology and is intended to attain equality of transmission opportunity and improvement of throughput. So there is a problem in that it is not suitable for RT communication that does not allow communication delay or retransmission.
There is prior art of increasing the diversity gain during retransmission based on line quality in a case where change over time of the propagation environment of radio signals is gradual (JP 2004-112098 A). However, during RT communication that does not allow communication delay or retransmission, this prior art does not lower the error rate to a quality that is allowed in RT communication, and does not make it possible to perform communication within a fixed period of time.